In wireless telecommunication systems information is transferred in an encoded form between a transmitting communication device and a receiving communication device. The transmitting communication device encodes original information into encoded information and sends it to the receiving communication device. The receiving communication device decodes the received encoded information in order to recreate the original information. The encoding and decoding is performed in codecs. Thus, the encoding is performed in a codec located in the transmitting communication device, and the decoding is performed in a codec located in the receiving communication device. However, since there are many different codecs available, the transmitting terminal and the receiving terminal have to agree upon the codec(s) to be used in a session. The selection of the codec takes place during the communication.
The GSM (Global System for Mobile Communication) codec mode selection over air interface is described next. The codec mode related information, which is transmitted on each link, contains CMI (Codec Mode Indication(s)) and CMC (Codec Mode Command(s)) in the downlink, respectively CMI and CMR (Codec Mode Request(s)) in the uplink. The CMI informs the receiver about the currently applied codec mode. The CMC informs the other end about the codec mode to be applied on the other link. The CMR informs the other end about the preferred codec mode on the other link. In the GSM, the codec mode information is transmitted in the speech traffic channel, using a part of its transmission capacity. Codec modes are constrained to change only every second speech frame. The CMCs/CMRs and the CMIs are altered such that they occur only every second frame. For codec mode adaptation the receiving side performs link quality measurements of the incoming link. The measurements are processed yielding a Quality Indicator (QI). For uplink (UL) adaptation, the QI is directly fed into the UL mode control unit. This unit compares the QI with certain thresholds and generates, also considering possible constraints from network control, the CMC indicating the codec mode to be used on the uplink. The CMC is then transmitted in the speech traffic channel to the mobile side where the incoming speech signal is encoded in the corresponding codec mode. For downlink (DL) adaptation, the DL Mode Request Generator within the mobile compares the DL Quality indicator with certain thresholds and generates a CMR indicating the preferred codec mode for the DL. The CMR is transmitted in the speech traffic channel to the network side where it is fed into the DL Mode Control unit. This unit generally grants the requested mode. However, considering possible constraints from network control, it may also override the request. The resulting codec mode is then applied to encoding of the incoming speech signal in downlink direction. Both for uplink and downlink, the presently applied codec mode is transmitted in the speech channel as CMI together with the coded speech data. At the decoder, the CMI is decoded and applied for decoding of the received speech data. In both UL and DL, there is always a transcoder in the network. The transcoder causes delays in the communication. Disadvantageously, the codec mode selection is only based on the quality of the radio interface.
The communication of the encoded information is critical for error free data communication in real-time applications such as a voice call. For example, in the voice call it is more preferable to use lower bit rate such as a lower codec mode with fewer errors than higher bit rate with larger number of errors. Generally, the communication of the real-time application uses lower bit rates with few data errors rather than high bit rates with data errors. The errors are due to packet losses or bit errors. Therefore, the selection of the codec is an important compromise between the data speed and QoS (Quality of Service).
One solution to provide feedback on the quality of the data distribution is an additional companion protocol, RTCP (Real-Time Control Protocol) operating in VoIP (Voice over Internet Protocol) systems. The transmitting communication device can make use of RTCP information to adapt the applied encoding scheme to changes in the network load in order to improve service at the receiving communication device. This requires that the devices support the RTCP that is undesirable because the devices would require more processing power and memory. The increase in required processing power leads to higher power consumption which is undesirable in wireless user terminals operated by a battery. Because the RTCP information needs to be communicated in backward direction, the communication of the RTCP information reserves and reduces network capacity from the actual services. One solution to reduce the network capacity is a header removal technique which is a method where the RTCP can be separated from the actual data. Thus, the actual data stream runs separately from the RTCP information. If the header removal is applied to the data stream, the RTCP needs to be run on a parallel PDP (Packet Data Protocol) context. However, the header removal technique requires additional mechanisms to link or create the removed header to the ‘header removed data’. Therefore, a substantial associative problem to link the data and the header emerge when the header removal of the packet is used because a recognition whether a packet is the RTCP packet or not is very difficult. There are other packets with or without the header in the data stream. Thereby, substantial difficulties emerge again in the linking. There is a need to observe the quality of the entire connection between the transmitting and receiving communication device and based on the observed quality select a communication mode depending on the quality.